1. Field of the Invention
The present invention relates to a method for an analysis of the transmission quality—frequently referred to in the literature as Quality of Service, QoS for short—in a packet-oriented network—especially an IP (Internet Protocol)-based network.
2. Description of the Related Art
As a result of an increasingly global alignment of companies, the use of telecommunication services to transfer voice and data is constantly increasing. The result of this is that the costs caused by these telecommunication services are constantly rising and are becoming a significant cost factor for the companies, which are looking for opportunities to reduce these costs. Local and global computer networks such as an ‘Intranet’ or the ‘Internet’ provide one option of transferring data cheaply and all around the world. In such cases real-time critical data, for example voice and video data, is increasingly transferred over these types of local and global packet-oriented networks.
For transfer of real-time critical data over packet-oriented networks there are not as a rule any constant transmission conditions—also frequently referred to in the literature as isochronous transmission conditions, because of a combination of the widest variety of influencing factors, such as the network used, the transmission protocol used etc. The result of the changing, i.e. non-isochronous, transmission conditions is an adverse effect on the quality of the transferred real-time data. One option for compensating for non-isochronous transmission conditions is the provision of so-called receive memories—also known in the literature as receiver buffers—which allow buffering of the received data and subsequent isochronous reading of the data. The setting of a receive buffer is in this case always a compromise between the highest possible compensation for non-isochronous transmission of the real-time critical data—and thereby a large receive buffer—and a minimum possible delay to the transmitted data—and thereby a small receive buffer.
A known method for determining the transmission quality in a packet-oriented network is what is known as the “ping” method in accordance with the ICMP (Internet Control Message Protocol) protocol, with which a test packet is sent and the receive delay is determined in the network by the response packet received—frequently known in the literature as the delay. A disadvantage of the ping method is that conventional data packets and not data packets for the transfer of real-time critical data—frequently referred to in the literature as VoIP (Voice over Internet protocol) packets are used for determining the transmission quality. As a result of different transmission mechanisms and thereby also of a different transmission behavior for real-time critical data and non-real-time critical data packets in the network, the special conditions of isochronous transmission cannot be sufficiently taken into account with the known method.
Furthermore, it has already been proposed in German Patent Application Number 102 10 651.7 that receive times of RTP (Real Time Protocol) data packets transferred in one direction are determined and so-called intermediate packet times calculated by continuous subtraction of the receive times of two consecutive data packets in each case. The sequence of intermediate packet times be used in a simple way to detect whether the data packets will be transported isochronously, or non-isochronously.
A further variable for the transmission quality in packet-oriented network is the number of data packets lost in data transmission in relation to the number of data packets transferred or expected at the recipient. Such a value is provided by default in the RTCP (Real Time Control Protocol) Standard by what is known as the “fraction lost field”.
A disadvantage of the method used within the context of the RTCP Standard however is that the time component of data transmission is not taken into account when determining the lost data packets. Thus with the method in accordance with RTCP Standard checking as to whether all previous data packets have been received is only undertaken on the basis of the last data packet received. No check is made on whether the data packet corresponding to a particular point in time was actually received at this time. If a connection is completely interrupted at a particular point, this fact is not fully reflected by the information contained in the “fraction lost field” (see FIG. 3 of the exemplary embodiment).
In particular a problem arises with send units that use what is known as “Voice Activity Detection”. With this method a send unit detects that there is not currently any voice data to be sent and sends what is known as an SID (Silence Insertion Descriptor) packet. As a rule data packets containing voice data are only sent again when there is voice data present.